Connect Mobile SIP application to GrandStream UCM6304A


Mahir never indicated which (webRTC or SIP) he would use. He specifically asked “For this, should I implement SIP or WebRTC protocol?”.

However, the GitHub project indicates -

  • Use pure dart-lang
  • SIP over WebSocket (use real SIP in your flutter mobile, desktop, web apps)
  • Audio/video calls (flutter-webrtc) and instant messaging
  • Support with standard SIP servers such as OpenSIPS, Kamailio, Asterisk and FreeSWITCH.
  • Support RFC2833 or INFO to send DTMF.

Which to use is really a choice based on what is to be accomplished. On the one hand, Mahir indicated his app would be used inside a closed wifi ecosystem, but then went on to indicate the app would also do nav, mapping, etc., in addition to making calls. I am not sure what or how the two indications jive with each other, but I do not profess to know what the goal is in any detail.

However, I tend to agree with Telcomsolutions in that SIP seems to be the path needed. WAVE is a closed system and one can always get an SBC to address the security issue.

If chat, video or other functionality is needed, then webRTC may be the one better suited. Mahir never stated the latter being required, only to make calls.


Dear users,

Thank you for all your feedback! If the app is depending on WSS + SIP, it should allow users to register. I have reported this question to our development team and I will let you know if we have any feedback. Thanks for your testing!

Thank you!


Hi Msr. Ipneblett,

I am thankful to you because you get me. telecom is a brand new domain for me. I might asked the question on wrong way.

I have my own closed LTE network and my application is running on this network. My client asked me to implement new feature to be able to make/receive a call (no video-no instant message) to/from any telephone.

For this problem we bought GrandStream. we attached PBX to GS and I created extension all my client in GS and PBX as well. I proofed this concept with 3th part application (Zoiper). I made a call, received a call everything works fine.

Simply, I am not sure which protocol should I implement to transfer a call from local network to PBX?

PS : many thanks to all of you,
Best Regards


SIP is, as Telcomm pointed out, an open standard and as he further indicated, the one most likely easier to implement and does accommodate the expressed need of being able to make/take calls.

The other concerns expressed about its use is the security issue and how being SIP, there are many hackers that will, if the port is exposed to the open Internet, will attempt to exploit it so that they can make calls at your expense.

SIP is governed by RFC3261 and then other standards that define how the protocol is implemented based upon the function being executed.

Good luck


@lpneblett Actually SIP is not a standard - it is just a protocol.


Correct, It is both a protocol and governed by a standard, which was defined by the IETF with RFC 3261-
SIP: Session Initiation Protocol

Status of this Memo

This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the “Internet
Official Protocol Standards” (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.

You can either be compliant or not, but being compliant is the standard by which all compliant devices are defined to be capable of using the protocol to interact with one another correctly.

As it had seemed that the question of which protocol had been resolved, I only pointed out the RFC as a reference along with an awareness that there are other RFCs that come into play.


Thanks you all for sharing these comments and suggestions with us!

We will save your comments and share with other users if they have the same concern. Thanks for all your testing!

Thank you!